Audio In Houses Of Worship
A comprehensive guide that will get you started in the world of church sound, no matter your level of familiarity.
Audio is an essential element in any modern-day religious service.
What is heard by the congregation is a combination of the acoustic qualities of the room and the performance of the audio system.
Some of the desirable acoustic qualities in a house of worship are:
Reverberance: When well controlled with early decay, the effect is perceived as a beautiful sound that enhances the quality of the audio. See the Rane Pro Audio Reference for a definition of “reverberation.”
Clarity: The ratio of the energy in the early sound compared to that in the reverberant sound.
Early sound is what is heard in the first 50 - 80 milliseconds after the arrival of the direct sound. It is a measure of the degree to which the individual sounds stand apart from one another.
Articulation: Determined from the direct-to-total arriving sound energy ratio. When this ratio is small, the character of consonants is obscured resulting in a loss of understanding the spoken word.
Listener envelopment: Results from the energy of the room coming from the sides of the listener. The effect is to draw the listener into the sound.
Where a conference room would be optimized for articulation and clarity, a symphony hall is optimized for reverberance and listener envelopment.
A good house of worship is optimized as a compromise between the somewhat conflicting requirements of music performance and the spoken word.
Articulation must be excellent but sufficient reverb is required to complement music performances. All reflections must be well controlled to achieve this balance and ensure the best possible listener experience.
An Example Of Good Sound
There are other possible examples but the author really likes this one. In some mosques, cathedrals and tabernacles there are wonderful domed ceilings that have marvelous natural acoustic properties.
The acoustic coupling from performers to the congregation grouped under the dome makes for a very (dare I say) “spiritual” experience. For the purpose of this article, this level of performance is a “gold standard” to which other acoustic spaces will be compared in the search for improvements and recommendations.
The U.S.A. Pavilion at Florida’s Epcot Center makes for an interesting case study. There is a dome ceiling in the pavilion. Under the dome an eight-part acappella group called the “Voices of Liberty” performs. For those under the dome listening to the group, the sound is beautiful and inspiring. Moving out from under the dome, the “magic” is gone.
This level of performance is not feasible in a typical house of worship but it does establish an icon as to what could be if there was sufficient skill (and budget) applied to the acoustic and audio system design.
And Now The Ugly World In Which We Live
Contrast this to a typical public address system squawking bad sound to the congregation.
That which was good is replaced with misery. You reach for a bottle of aspirin to calm the headache induced by a pair of blaring powered speakers.
Some of the problems encountered by audio designers/consultants include:
Excessive Reverberation—such that articulation and clarity is poor.
Echo—where a discrete sound reflection returns to a listener more then 50 milliseconds from the direct sound and is significantly louder then the reverberation sound.
Flutter echo—repeated echoes that are experienced in rapid succession that occur between two hard parallel surfaces. All echoes ruin the acoustic properties of a room and a flutter echo is particularly damaging.
Coloration due to reflections—when a reflection destructively recombines with the direct sound modifying the frequency response in the process. These are non-minimum-phase colorations as correction with equalization is not possible.
Delayed Sound—from coupled volumes (contamination from adjacent rooms storing sound energy and then returning the energy to the main room).
Psychological preconditioning—It is a common problem for the clergy and congregation to be so preconditioned by bad sound that they become resistant to change and find it difficult to (at first) recognize good sound.
Figure 1. Microphone to Amplifier Chain.
This can also work in the audio consultants favor when the customers are preconditioned by good sound and are willing to invest the required resources toward good audio design.
For those of us designing audio for houses of worship with a rectangular room, flat walls and probably a vaulted ceiling, some form of sound reinforcement is required. Through attention to detail and careful design of the audio system, the experience of the congregation can be non-aspirin inducing and the system simple to use.
Common Signal Processing Blocks
Let’s begin by looking at the universal signal processing chain common to all audio systems. In the simplest systems these functions are accomplished in an audio mixer that feeds a pair of powered speakers.
More sophisticated systems include equalization, compression, limiting, automation, feedback suppression, electronic crossovers and other tools of the trade. These days it is possible to include all of these functions in a DSP (Digital Signal Processor). One example of the signal chain from the minister’s microphone to the power amplifiers is shown in Figure 1.
The signal processing flow starts at the Analog Input. A 2-band Parametric Equalizer filters out-of-band low frequencies. The microphone signals are summed together in an Automatic Mixer. An AGC (Automatic Gain Control) reduces the dynamic range and a High-Pass Filter in the side chain improves the performance of the AGC.
The Level control can be tied to a pot on the wall or a smart remote. There is a Feedback Suppressor for good measure. A 2-way Crossover supports a biamplified system. The 10-band Parametric Equalizers are utilized for both wide- and narrow-band corrections.
Generally, wide-band filters correct minimum-phase frequency response irregularities in the speaker drivers and in the room response. Narrow-band filters are useful to partially correct non-minimum-phase related problems such as energy stored in room modes (reverberant energy).
A Limiter could also have been added to protect the system from clipping if that feature is not included in the power amplifier.
Now let’s take a look at some of these signal processing blocks in greater detail.
Analog Input / Microphone Preamp
It is surprising how often even experienced audio consultants will configure an audio input incorrectly.
It is important that as much gain as possible is accomplished at the front end of the system in the Analog Gain stage.
Any additional gain from Digital Trim after the input stage degrades optimum signal-to-noise performance.
As an example, let’s set the input gain to a value of +40 dB.
One way is where the analog gain is set to a value of +45 dB and the digital trim is set to -5 dB (as in Figure 2), the measured input referred noise is -127 dBu.
Figure 2. Drag Net Input Block.
A common (but incorrect) way would have the analog gain set to a value of +30 dB and the digital trim set to +10 dB (the author has seen this repeatedly), to give the same Mic gain of 40 dB—but now the input-referred noise is degraded to -114 dBu.
That is an increase of 13 dB for the noise floor, or a change (in the bad direction) of 8 dB in the maximum SNR (Signal to Noise Ratio). Your exercise is to determine why the SNR was only degraded by 8 dB rather then the intuitively obvious value of 13 dB.
Answer: The noise floor does drop by 13 dB, but this combination of settings causes the analog input stage to clip at an input level that is 5 dB lower. Hence, the change in system SNR is 8 dB.
Figure 3. Drag Net Parametric for Input Low Cut.
Applying attenuation after the input stage (rather then gain) reduces overload performance and so should be used with skill and discretion. It is the proper technique to maximize noise performance.
Input Low-Cut Filter
A very good idea is to add a low-cut filter set to ~80 Hz after the input stage to minimize the effects of undesirable low-frequency noises such as bumps and thumps that come from handling the mic and also wind blasts and pops from speaking too closely into the microphone.
Figure 4. Drag Net Parametric for AGC Side Chain.
In Figure 3, both 2nd-order filters are set to the same frequency to produce a 4th-order filter.
There should also be a low-cut filter in line with the SC (Side Chain) input of the AGC (Automatic Gain Control).
This filter can be set to a higher corner frequency (such as 120 Hz in Figure 4) to improve the performance of the AGC by rejecting the effects of low frequency noises.
The Auto Mixer—A Little Automation Buddy
An Auto Mixer (shown in Figure 5) is a good idea when there is more then a single open microphone.
Auto Mixers combine the signals from multiple microphones and automatically correct for the changing gain requirements as the NOM (Number of Open Microphones) changes.
Figure 5. Drag Net Auto Mixer Block.
Threshold with Last On is a useful setting for all microphones used in a worship service (Figure 6).
Unused microphones (input levels are below threshold) are gated. When the input of a microphone is above threshold then other inputs with a lower assigned priority level are ducked.
Figure 6. Auto Mixer Input Edit Block.
Automatic Gain Control
A Compressor is the correct processing block in this link of the audio chain. Something is needed here to prevent exuberant preaching from melting down the congregation.
Surprisingly, an AGC can be very useful in this position but configured to behave more like a specialized compressor by using the settings shown in Figure 7.
Figure 7. Drag Net AGC Block.
The value of “Threshold re: Target” is set to have an offset of 0 dBr so that “Threshold” has the same value as the “Target.” “Maximum Gain” becomes 0 dB and the gain curve starts to look like a compressor but there are additional controls in an AGC for Hold and Release that are useful when the input level is below threshold.
These settings avoid the problems of compressor “pumping” when that exuberant speaker is at the microphone as attenuation levels are held between spoken phrases.
Then, when transitioning to a more reserved speaker, the hold time (below threshold) is short enough to expire so that the gain returns to a normal level.
An Exciting Labor-Saving Tip—Put a Control On the Wall
Here is an exciting tip. A level control can provide attenuation as needed under the control of a pot on the wall or a smart remote.
This is handy in systems where a minister needs to run a system alone without the assistance of an audio specialist who is running a mixing board. The remote can be located on or close to a pulpit which places control of the audio system at the fingertips of the minister. The DSP control is shown in Figure 8.
Figure 8. Drag Net Level Block Mapped to a Remote Level Control.
Feedback Suppression—A Gift From Above?
The next item in this processing chain is somewhat controversial. It is a Feedback Suppressor.
To some audio consultants a Feedback Suppressor is heresy! The argument is that a properly calibrated system has no need of such a Band-Aid.
This is generally true, but there is one case when it is wise for an audio consultant to suffer the ignominy of using a Feedback Suppressor—a lay clergy where the person speaking is untrained and/or unfamiliar with proper use of a microphone.
Figure 9. Drag Net Feedback Suppressor.
The author has witnessed such a person cup their hands (in the attitude of prayer) directly around the microphone capsule. The hands form a resonant chamber that results in squealing feedback.
A good Feedback Suppressor would have locked on to the offending tone and notched it out posthaste.
Parametric Equalization: Now We’re Having Real Fun
Parametric equalizers are used for both wide and narrow band corrections.
Generally, wide-band and shelf filters can correct for minimum-phase frequency response irregularities.
One interesting detail of Figure 10 is Hi-Shelf Filter 1. This filter was added after achieving flat in-room response.
Since the system was calibrated in an empty room, this extra high-frequency energy is intended to compensate for the high-frequency absorption of the congregation when the room is full of people.
There is also a noise-masking effect in some congregations that will tend to obscure the intelligibility of the spoken word. In practice this approach of adding a bit of extra high-frequency energy into the room works well.
Figure 10. Drag Net Parametric Block (May Have up to 15 Bands per Block).
Narrow-band filters (see Figure 11) are useful to partially correct non-minimum-phase related problems such as energy stored in room modes.
At low frequencies this energy causes bass to sound indistinct, and in midrange to lower treble this energy is perceived as reverberation.
These filters attenuate the frequencies that bounce about the room. In an acoustically live room, room resonances can propagate for a surprisingly long time causing these frequencies to “build up.”
Figure 11. Parametric with Narrow-Band Filters.
Narrow-band filters are just a partial solution. Greatest effectiveness is achieved when filters are used in conjunction with acoustic room treatments such as diffusers, high/mid frequency absorbers and bass traps.
Specific Examples
Example #1: A Small Church
Description:
The ceiling is low suspended acoustic tile over an open space covered with thin carpet. The RT60 (the time it takes the reverberant sound to decrease by 60 dB) is short, so controlling reverberation is not a problem for audio clarity.
In fact, the room is a touch “dry” for music, and content of the worship service includes live musical performances.
The sources of audio are the minister with a wireless microphone and the band.
Additional sources are DVD/CD players and other devices as needed. Control is via a 24-channel mixer with all inputs used.
Output is to a pair of powered speakers mounted high in the corners of the room in a stereo configuration. This installation was done by members of the congregation without consultation with an audio professional.
Next, let’s look at some specific examples to bette illustrate these points.
Problems:
• The quality of the audio is poor with numerous problems including uneven frequency response.
• An experienced sound person is required to run the mixer for all audio system use.
• There is poor coverage of the congregation from the stereo speaker pair. People sitting in the hot spots just in front of the speakers are blasted with excessive level, and the rest of the congregation is exposed to a strong interference pattern between the two speakers.
Figure 12. Stereo Speaker Pair Coverage.
The system is uncompensated for room modes, room response and speaker response irregularities.
There is a small “sweet spot” in the center of the room where the two speakers combine coherently but there is an isle down the center of the seats. Since there are no chairs, no one is seated in the “sweet spot”.
So does this audio system work the way it is? Yes, but even the pastor knows the congregation may not be receiving the best possible audio experience. This example is rich in possibilities.
Recommendations
Improvements to this system are accomplished in a number of ways. A DSP can be used for equalization, other processing and to add automation to the minister’s microphone.
The entire worship band could be run through a mixer with each individual input processed by an AGC.
There are admittedly downsides to automating the audio mixing of a large group, as the automation is not as intelligent as an experienced sound person, but is possible in some cases.
The speaker system is examined to look at options that provide more even coverage of the congregation. Improvements to this audio system can be introduced in phases.
Phase 1:
Add a DSP box between the output of the mixer and the feeds to the main speakers and on-stage monitors. Features added could be:
• Parametric Wide-Band Equalization. This alone would greatly improve this system.
• Parametric Narrow-Band Equalization. A short RT60 makes this unnecessary at this time. However, remodeling could increase RT60 to where narrow-band equalization would be needed. (This room could use bass absorbers).
• High-Pass Filtering. If not in the 24-channel mixer already.
• Compression. Always a good idea with microphones because of the inverse square law relationship between the preacher’s mouth and the location of the microphone.
• Feedback Suppression. If needed.
Phase 2:
Automation is incorporated with automixers and remote controls. There are many exciting ways to add these features depending on the needs of individual congregations.
The most obvious upgrade would be to add the ability for a minister to turn on and control the main microphones from a simple control panel located in easy reach at the front of the room.
Phase 3
The very uneven coverage of the congregation by the stereo speaker pair needs to be addressed, as shown in Figure 12. The seats directly in front of the speakers have enough level to kill small animals.
If the audio system were perfect then each seat in the congregation would have the same audio level. In the author’s experience, similar rooms have been controlled within a couple of dB.
In this example, the seat closest to each loudspeaker is about 15 dB louder then the worst seat on the floor, and interference between the two speakers adds to a very lumpy and unpleasant frequency response.
Another problem is that the FOH (Front Of House) Mixer is placed in a location for good sound, causing the levels at the ends of the front rows to be way too loud.
Line Array Speakers
One improvement is to remove the stereo pair of point-source loudspeakers, and install a floor-to-ceiling line array located in the center of the back wall as shown in Figure 13. Coverage of the congregation is more even, and the level at the FOH Mixer location is very similar to the coverage level over the whole floor of the congregation.
The level of the stage monitors is greatly reduced and some of the stage monitors may no longer be needed depending on the individual needs of the musicians.
Figure 13. Line Array Speaker Coverage.
Within the near field of the line array there is a range were the audio level will decrease by only 3 dB for each doubling of distance which greatly helps even the coverage across the entire floor.
One other characteristic of this application is that the audio is distributed across the whole line so that even if a microphone is right next to the line there is little tendency to feedback.
In this example, there is a low suspended-acoustic-tile ceiling that shortens the length of a line array speaker. This limits some of the good qualities of a line array so this might not be the best solution.
If the room were remodeled so there was a high ceiling, then a line array would make more sense because a longer line array would fit. This is especially true if the newly remodeled ceiling was acoustically reflective causing the RT60 of the room to be much greater.
The high directivity of a long line array greatly helps to project the audio out to the floor rather then have the audio directed toward the ceiling where it contributes to the reverberant energy and slap echoes in the room.
Supplemental Distributed Array Speakers
Because of the dropped ceiling, another option would be a distributed array of supplemental ceiling speakers in the back of the room as shown in Figure 14. The loudness level of the main stereo pair could be reduced by at least 12 dB.
This would greatly diminish the effects of the hot spots in the front of the room but would leave the level at the back of the room way too low. Ceiling speakers can be added in the locations shown to fill in the audio in the back of the room.
It would be very important to include a speaker over the mixer location so the audio at that location matches the level in the congregation to aid in achieving an accurate mix.
Why The Delay?
The ceiling loudspeaker signals should be delayed in time so their output combines coherently with the output from the point-source pair in the front of the room.
Figure 14. Distributed Array Speaker Coverage.
If the rear loudspeakers are not correctly delayed then the loudspeakers in the room will not combine correctly.
This room is too small for audio from the front of the room to be perceived as a distinct echo.
Applying a proper delay to the ceiling speakers can minimize the problem of localization confusion that occurs if the first arrival sound is coming from the overhead loudspeakers and not the front of the room.
Example #2: A Mid-Sized Contemporary House of Worship
Description:
This second example is a medium sized house of worship. The vaulted ceiling is high and the floor in the congregational seating area is covered with hard-industrial vinyl.
The RT60 is longer then the first example at approximately 1.5 seconds so reverberation is a problem in an empty room. The sources of audio are again ministers on a microphone and a worship band.
Control is via a 32-channel mixer. The speaker system is an array of three large boxes mounted as a central cluster high in the peak of the ceiling. A professional audio company did the installation and calibration of the audio system.
The quality of the audio in this church is much better than in the first example. An interesting question is: how good is “good enough”? When interviewed, members of this congregation can usually hear. Rarely is the audio painful to listen to so some say that the audio quality is fully acceptable.
This is a good time to reflect back on the example in the introduction where domed ceilings were held up as an icon of natural acoustic wonderfulness. Let’s examine each individual audio characteristic previously discussed and see how this audio system installation stacks up.
Problems
• Reverberance is not well controlled and is dependent on the configuration and occupancy of the room. Low-mid frequencies are a particular problem as the energy builds up and is never trapped or controlled.
• Clarity is fairly good and meets a minimum standard.
• Articulation is acceptable but not outstanding. The ALCONs (Articulation Loss of Consonants) rating of this room is fairly low but in the acceptable range. However, there is room for improvement.
• Listener envelopment is nonexistent and completely pales in comparison to the example of a domed ceiling.
• Again, as in the first example, an experienced sound person is required to run the mixer for any use of the audio system, as there is no automation in the audio system.
• There is good coverage of the congregation from the central cluster, but people sitting in the area where the coverage patterns between two of the speakers overlap experience uneven frequency response due to the comb filtering caused by the interference between these two speakers.
• Bass response is particularly poor. The poor bass response leads to the impression that the system lacks sufficient power.
Recommendations
A DSP is already in the system and can be used for additional equalization and other tasks. The same recommendation applies to add enough automation so that a simple service can be done without bringing in a sound person.
The speaker system may already be fully adequate.
Figure 15. Distributed Array Speaker Coverage.
The first temptation may be to add a subwoofer to add bass power, but after a quick survey it is probable that the buildup of mid-bass energy in this room makes the quality of the bass so poor that adding more bass will only make matters worse.
To fix the room, the ceiling and walls could be completely covered in bass absorptive panels, but this is not really practical so a compromise is to add bass traps to the corners of the room and the ridge of the ceiling.
If it is not possible to tame the room with traps, then narrow-band filtering techniques could be employed.
This is where the room is evaluated for the natural modes that build up energy in the room and these frequencies are notched out with a very narrow filter. A combination of some absorptive panels and narrow-band filters might be the best compromise.
There are regions (as shown in Figure 15) where the coverage from the individual speakers in the cluster interfere with each other rather than combine cooperatively. This interference is frequency-dependent.
The solution is to reduce the contribution of some of the speakers of those problem frequencies so that interference is minimized.
The system would then require re-calibration to complement the above changes. That should do it.
A the time of publication Michaël Rollins was a senior digital design engineer for Rane Corporation.
Download a copy of this article. (pdf)
Editors Note: This and other educational articles are available in the RaneNote Library, a subset of the Rane ProAudio Reference.
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For The Record: The Past Tells Us Much About The Future Of Live Recording
We should always remember to look back at the historical trends of our industry - it’s the only way we can stay ahead of the curve and keep providing the gear and the services that our clients need
Many of us make our livings providing concert-goers with the best live music experience possible. We deploy high-fidelity loudspeaker systems and microphones with the latest in digital effects and studio-quality processing in an effort to make the live show sound “just like the record.”
Only better, of course, because the excitement, visual elements, crowd response and performance spontaneity are impossible to reproduce in someone’s living room. Or is it? Let’s step back in time and examine our progress in the effort to capture the live experience for the fans to take home.
Live recording has taken many forms over the years. In the big band era, it was typical to put a single microphone out in front of the performers and hope for the best. Early refinements consisted of adding a second mic for the soloists to step up to. Performances were largely acoustic, with the possible exception of a lead vocalist, so there was no interface with the live reinforcement system.
Recordings were monophonic, and the only options available to the recordist for influencing the outcome were mic choice and location. By the way, the delivery system was usually 78 RPM vinyl records. Given the limitations, it’s amazing how many vibrant, exciting examples exist from that era of music.
Through the 1950s and 60s, there were huge changes in performance, recording and playback technology. On the performance side, the invention of the electric guitar changed everything. (In fact, a case can be made that the electric guitar spawned our entire industry.) The concept of an amplified performance where the audience heard an electronic representation of the instrument rather than the instrument itself was revolutionary in many ways.
It wasn’t long before the bass joined the ranks of amplified instruments, and all of the other musicians (with the possible exception of the drums) were using mics. This allowed shows to be staged at much larger halls than was possible in the “acoustic era,” enhancing exposure for the artist - and revenue for everyone. I
This also shrank the size of performing groups as well. Previously, if the trombones needed to be louder, more trombone players were added. Now the trombone sound could simply be turned up.
THE MIGRATION
On the recording front, the big news was multiple tracks. Two- and even three- track recorders were invented. This created a need for mixing consoles, and most were built by the studio owners themselves. Some even sported advanced features like equalization.
This technology then migrated over to sound reinforcement, and it required operators. We all got a job!
Big things were happening on the playback scene as well. The hi-fi craze swept many parts of the world. Playback systems with wide frequency response and low distortion became available. The 33 RPM Long Play (LP) record allowed much longer playing times.
Meanwhile, stereophonic sound finally gave recorded music more of the spatial impact of a live performance. With stereo playback, the instruments could be spread across the soundstage to simulate sitting in front of a real band. The elements required to bring the live performance experience into the listener’s home were falling into place.
As the music business roared into the 1970s, the capability grew to duplicate the recording techniques for live events. Record companies wanted to be able to issue as many LP’s as possible from their hottest bands. One way to do this - without taking them off the road – was the live album.
As budgets became available for quality live recording the first studio trucks were created. A recording studio control room was crammed into a box truck and trundled off to the gig. Either using splits off the sound reinforcement mics or double mic’ing everything. a quality multi-track recording an actual concert could be made. A few audience mics were added, and voila, the record company had their new release.
The best part? No new songs had to be written. The same songs could be sold to eager fans twice! Soon, no self-respecting band was without a live album. Of course, the other advantage was that if the house mix or sound system was substandard, or the acoustics were bad, a multi-track master tape provided some ability to “fix it in the mix.”
And on more than a few instances the band would nip into the studio to fix “green notes” in the vocals or a botched guitar lead.
Live recording had started to generate its own revenue stream, which supplemented the box office receipts from the show. Eventually someone got the bright idea of bringing a movie camera into the proceedings. Between the audio recording truck, the camera operators, directors and miscellaneous technical personnel, it could turn into a huge undertaking. For some events, it was worth the money.
The Woodstock movie made far more cash than the festival itself. If you couldn’t go to the concert, the concert would come to your local movie theater. But only the biggest bands or the most high profile events could justify the expense of the production and pack the fans into theaters.
INNOCENTLY ENOUGH
As technology continued it’s relentless march, many acts wanted to record every performance. It started innocently enough with the ubiquitous “board tape.” At first this was just a stereo cassette coming right off the same stereo pair feeding the mains. These tapes were generally used by the band and their management to review the night’s performance.
Of course, sometimes this led to some mix criticism as well. It was hard to explain to a guitar player that the reason he couldn’t hear himself on the board tape was because his stage amplifiers were on “11” and his mic was off.
So eventually we started doing sub mixes for the board tapes. I’ve done tours where I had a combination of pre-fader and post-fader stereo aux sends, and used delays to time align an X-Y stereo pair of room mics into a DAT machine – all just to make the troops happy with their review tapes.
And inevitably some bright soul would say, “We could release this as a live album!” or maybe give their copy to their girlfriend, which later appeared as a bootleg causing great consternation and finger-pointing within the ranks. But that’s another story.
I saw one act that even carried a 24-track recorder in a huge flight case and a maintenance technician on tour so they could record every night. They even organized their set list to give the tech time to change tapes. A sound company I worked for owned a Midas Pro 5 board reputedly built for Harry Belafonte (and of course christened the “Day-O” board), and it had an extra 24 output buses to feed his recorder. It also weighed a ton.
But once again technology came to the rescue.
In the 1990s, digital recorders utilizing tape cartridges were introduced. Each unit recorded eight tracks and several could be synched up. They were rack mountable, reasonably light and low maintenance. A portable rack could now hold enough recorders to run a direct out from every board channel and record every night for future use.
Some enterprising engineers even used the previous night’s show routed back to the console to do a preliminary sound check. The only downfall was that you had to spend every spare moment formatting tapes for the recorders, and archiving was a pain. Depending on the length of the show and the number of tracks required, a single performance might use 30 tapes or even more.
By this time, almost every home had at least a decent stereo and a VCR. More and more tours were filmed, whether a theatrical release was realistic or not. Home entertainment technology had created an alternative market for video concert releases.
Although live records were still being released, the concert experience had much more impact if the visual elements were included. Most top tours and almost all major festivals had an audio and video recording element to document the event and provide a revenue stream long after the actual show. The concert experience was now as close as your local video store.
COMBINATION OF FORCES
The 21st Century has only expanded this paradigm. A combination of forces has created a “perfect storm” supporting concert recording. On the recording technology front, digital audio workstations are smaller, lighter, more robust, and in fact, are often the same machines being used in the recording studio.
An entire show can be recorded on a single hard drive. Digital consoles can easily provide audio streams to the recorders without multiple analog to digital (A-D) conversions or analog signal splits.
The advent of the DVD and home theater systems provide a delivery medium with the quality and impact to really bring the concert experience into the home. Large high-definition screens and surround sound can do a remarkable job of reproducing the feeling of being at an event. They also provide new ways to make money from a live performance, and in a day and age where file sharing and piracy have eaten away at the traditional money flow in the music business.
It used to be common for record companies to provide tour support from record sale receipts. Now, it’s more common for touring and the recorded products that come from touring to be the largest source of income for performers.
Some bands have taken it to the next level by selling recordings of the actual show to attendees on their way out. “Jam bands” are still popular, and no two performances are alike. So getting a recording of these performances show may have more significance than whether the band says, “Good night, Seattle” or “Good night, Detroit”. Concerts are being staged for the sole purpose of producing a DVD or even a pay-per-view broadcast.
A LONG TIME
Nothing can really replace the adrenaline, the excitement and the immediacy of being at a great concert. Our jobs are going to be around for a long time.
But we should always remember to look back at the historical trends of our industry. It’s the only way we can stay ahead of the curve and keep providing the gear and the services that our clients need.
And anything that enhances the revenue stream from live performances for the artists, promoters - and especially for us - is a very good thing indeed.
Bruce Main has been a systems engineer and front of house mixer for more than 35 years. He has also built, owned and operated recording studios and designed and installed sound systems.
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